blob: bd4340e7a189d167dadb6bd0a793e1f7e8e2d7fa [file] [log] [blame]
/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/dtmfsender.h"
#include <ctype.h>
#include <string>
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
namespace webrtc {
enum {
MSG_DO_INSERT_DTMF = 0,
};
// RFC4733
// +-------+--------+------+---------+
// | Event | Code | Type | Volume? |
// +-------+--------+------+---------+
// | 0--9 | 0--9 | tone | yes |
// | * | 10 | tone | yes |
// | # | 11 | tone | yes |
// | A--D | 12--15 | tone | yes |
// +-------+--------+------+---------+
// The "," is a special event defined by the WebRTC spec. It means to delay for
// 2 seconds before processing the next tone. We use -1 as its code.
static const int kDtmfCodeTwoSecondDelay = -1;
static const int kDtmfTwoSecondInMs = 2000;
static const char kDtmfValidTones[] = ",0123456789*#ABCDabcd";
static const char kDtmfTonesTable[] = ",0123456789*#ABCD";
// The duration cannot be more than 6000ms or less than 70ms. The gap between
// tones must be at least 50 ms.
static const int kDtmfDefaultDurationMs = 100;
static const int kDtmfMinDurationMs = 70;
static const int kDtmfMaxDurationMs = 6000;
static const int kDtmfDefaultGapMs = 50;
static const int kDtmfMinGapMs = 50;
// Get DTMF code from the DTMF event character.
bool GetDtmfCode(char tone, int* code) {
// Convert a-d to A-D.
char event = toupper(tone);
const char* p = strchr(kDtmfTonesTable, event);
if (!p) {
return false;
}
*code = p - kDtmfTonesTable - 1;
return true;
}
rtc::scoped_refptr<DtmfSender> DtmfSender::Create(
AudioTrackInterface* track,
rtc::Thread* signaling_thread,
DtmfProviderInterface* provider) {
if (!track || !signaling_thread) {
return NULL;
}
rtc::scoped_refptr<DtmfSender> dtmf_sender(
new rtc::RefCountedObject<DtmfSender>(track, signaling_thread,
provider));
return dtmf_sender;
}
DtmfSender::DtmfSender(AudioTrackInterface* track,
rtc::Thread* signaling_thread,
DtmfProviderInterface* provider)
: track_(track),
observer_(NULL),
signaling_thread_(signaling_thread),
provider_(provider),
duration_(kDtmfDefaultDurationMs),
inter_tone_gap_(kDtmfDefaultGapMs) {
ASSERT(track_ != NULL);
ASSERT(signaling_thread_ != NULL);
// TODO(deadbeef): Once we can use shared_ptr and weak_ptr,
// do that instead of relying on a "destroyed" signal.
if (provider_) {
ASSERT(provider_->GetOnDestroyedSignal() != NULL);
provider_->GetOnDestroyedSignal()->connect(
this, &DtmfSender::OnProviderDestroyed);
}
}
DtmfSender::~DtmfSender() {
StopSending();
}
void DtmfSender::RegisterObserver(DtmfSenderObserverInterface* observer) {
observer_ = observer;
}
void DtmfSender::UnregisterObserver() {
observer_ = NULL;
}
bool DtmfSender::CanInsertDtmf() {
ASSERT(signaling_thread_->IsCurrent());
if (!provider_) {
return false;
}
return provider_->CanInsertDtmf(track_->id());
}
bool DtmfSender::InsertDtmf(const std::string& tones, int duration,
int inter_tone_gap) {
ASSERT(signaling_thread_->IsCurrent());
if (duration > kDtmfMaxDurationMs ||
duration < kDtmfMinDurationMs ||
inter_tone_gap < kDtmfMinGapMs) {
LOG(LS_ERROR) << "InsertDtmf is called with invalid duration or tones gap. "
<< "The duration cannot be more than " << kDtmfMaxDurationMs
<< "ms or less than " << kDtmfMinDurationMs << "ms. "
<< "The gap between tones must be at least " << kDtmfMinGapMs << "ms.";
return false;
}
if (!CanInsertDtmf()) {
LOG(LS_ERROR)
<< "InsertDtmf is called on DtmfSender that can't send DTMF.";
return false;
}
tones_ = tones;
duration_ = duration;
inter_tone_gap_ = inter_tone_gap;
// Clear the previous queue.
signaling_thread_->Clear(this, MSG_DO_INSERT_DTMF);
// Kick off a new DTMF task queue.
signaling_thread_->Post(RTC_FROM_HERE, this, MSG_DO_INSERT_DTMF);
return true;
}
const AudioTrackInterface* DtmfSender::track() const {
return track_;
}
std::string DtmfSender::tones() const {
return tones_;
}
int DtmfSender::duration() const {
return duration_;
}
int DtmfSender::inter_tone_gap() const {
return inter_tone_gap_;
}
void DtmfSender::OnMessage(rtc::Message* msg) {
switch (msg->message_id) {
case MSG_DO_INSERT_DTMF: {
DoInsertDtmf();
break;
}
default: {
ASSERT(false);
break;
}
}
}
void DtmfSender::DoInsertDtmf() {
ASSERT(signaling_thread_->IsCurrent());
// Get the first DTMF tone from the tone buffer. Unrecognized characters will
// be ignored and skipped.
size_t first_tone_pos = tones_.find_first_of(kDtmfValidTones);
int code = 0;
if (first_tone_pos == std::string::npos) {
tones_.clear();
// Fire a “OnToneChange” event with an empty string and stop.
if (observer_) {
observer_->OnToneChange(std::string());
}
return;
} else {
char tone = tones_[first_tone_pos];
if (!GetDtmfCode(tone, &code)) {
// The find_first_of(kDtmfValidTones) should have guarantee |tone| is
// a valid DTMF tone.
ASSERT(false);
}
}
int tone_gap = inter_tone_gap_;
if (code == kDtmfCodeTwoSecondDelay) {
// Special case defined by WebRTC - The character',' indicates a delay of 2
// seconds before processing the next character in the tones parameter.
tone_gap = kDtmfTwoSecondInMs;
} else {
if (!provider_) {
LOG(LS_ERROR) << "The DtmfProvider has been destroyed.";
return;
}
// The provider starts playout of the given tone on the
// associated RTP media stream, using the appropriate codec.
if (!provider_->InsertDtmf(track_->id(), code, duration_)) {
LOG(LS_ERROR) << "The DtmfProvider can no longer send DTMF.";
return;
}
// Wait for the number of milliseconds specified by |duration_|.
tone_gap += duration_;
}
// Fire a “OnToneChange” event with the tone that's just processed.
if (observer_) {
observer_->OnToneChange(tones_.substr(first_tone_pos, 1));
}
// Erase the unrecognized characters plus the tone that's just processed.
tones_.erase(0, first_tone_pos + 1);
// Continue with the next tone.
signaling_thread_->PostDelayed(RTC_FROM_HERE, tone_gap, this,
MSG_DO_INSERT_DTMF);
}
void DtmfSender::OnProviderDestroyed() {
LOG(LS_INFO) << "The Dtmf provider is deleted. Clear the sending queue.";
StopSending();
provider_ = NULL;
}
void DtmfSender::StopSending() {
signaling_thread_->Clear(this);
}
} // namespace webrtc